SIP trunk pblm- Some numbers not connecting

[Eldho E Kuriakose]

Dear Expers

I am in a wred problem as some of the calls not connecting via sip gateway to PSTN. But when we are calling from mobile it is connecting

I have collected the logs and seems to be the IST issue. Can you please doublecheck

calling party:8307

called:0138425655 -working

called:0138425653 -not working

but from mobile both numbers ringing. The debug ccsip messages from router for the first and second call is attached herewith 

[Eldho E Kuriakose]

I have seen 

Reason: Q.850;cause=57;text="bearer capability not authorized"

and form my running config

voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729r8
 codec preference 4 g729br8

dial-peer voice 6 voip
 description ** National **
 translation-profile outgoing ADDTran
 destination-pattern 01[^0].......
 session protocol sipv2
 session target ipv4:10.200.7.157
 voice-class codec 1
 dtmf-relay rtp-nte
 no vad

Do I need to change codec transparent?

[Mohammed Khan · Accepted solution]

Hi eldho k,

From the logs i see that Telco is sending SIP 403 with Reason: Q.850;cause=57;text="bearer capability not authorized" You got to reach to Telco and find why Telco sends this.

Logs snips below

\\ SIP Invite \\

582258: Feb  9 15:36:36.229: //3140974/69BA18800001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0138425653@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.50.186:5060;branch=z9hG4bK902FE88E
Remote-Party-ID: "Madan Gopal Bonula" <sip:2903000@172.29.50.186>;party=calling;screen=yes;privacy=off
From: "Madan Gopal Bonula" <sip:2903000@172.29.50.186>;tag=5C2B05EC-9D9
To: <sip:0138425653@10.200.7.157>
Date: Mon, 09 Feb 2015 15:36:36 GMT
Call-ID: 443EDB18-AFA811E4-AB4CB164-B78FCB20@172.29.50.186
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1773803648-0000065536-0000096275-0167966730
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1423496196
Contact: <sip:2903000@172.29.50.186:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250

\\SIP Trying \\

582260: Feb  9 15:36:36.241: //3140974/69BA18800001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.29.50.186:5060;branch=z9hG4bK902FE88E
Call-ID: 443EDB18-AFA811E4-AB4CB164-B78FCB20@172.29.50.186
From: "Madan Gopal Bonula"<sip:2903000@172.29.50.186>;tag=5C2B05EC-9D9
To: <sip:0138425653@10.200.7.157>
CSeq: 101 INVITE
Content-Length: 0

\\SIP 403 \\

582261: Feb  9 15:36:36.449: //3140974/69BA18800001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.29.50.186:5060;branch=z9hG4bK902FE88E
Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
Call-ID: 443EDB18-AFA811E4-AB4CB164-B78FCB20@172.29.50.186
From: "Madan Gopal Bonula"<sip:2903000@172.29.50.186>;tag=5C2B05EC-9D9
To: <sip:0138425653@10.200.7.157>;tag=sbc0802pduuhubt
CSeq: 101 INVITE
Reason: Q.850;cause=57;text="bearer capability not authorized"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0

[Eldho E Kuriakose]

Will a transcoder configuration in CUCM will resolve this issue?

[Mohammed Khan]

Hi eldho k,

Transcoder config wil not help. Its a telco issue.

Regards,

Mohammed Noor

[Eldho E Kuriakose]

The similar issue mentioned here

https://supportforums.cisco.com/discussion/12051436/outbound-call-failing-cause-code-57 

reslved by configuring transcoder

[Mohammed Khan]

Hi eldho k,

I read the other thread, i see it resolved by configuring transcoder because telco was supporting G729.

Per the SIP Invite of your sample call we Router sends G711ulaw. if codec was the issue telco should have responded back with SIP 488 Not Acceptable Here. I am pretty sure adding transcoder will not help. how ever you can give it a try.

582258: Feb  9 15:36:36.229: //3140974/69BA18800001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0138425653@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.50.186:5060;branch=z9hG4bK902FE88E
Remote-Party-ID: "Madan Gopal Bonula" <sip:2903000@172.29.50.186>;party=calling;screen=yes;privacy=off
From: "Madan Gopal Bonula" <sip:2903000@172.29.50.186>;tag=5C2B05EC-9D9
To: <sip:0138425653@10.200.7.157>
Date: Mon, 09 Feb 2015 15:36:36 GMT
Call-ID: 443EDB18-AFA811E4-AB4CB164-B78FCB20@172.29.50.186
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1773803648-0000065536-0000096275-0167966730
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1423496196
Contact: <sip:2903000@172.29.50.186:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 6507 1616 IN IP4 172.29.50.186
s=SIP Call
c=IN IP4 172.29.50.186
t=0 0
m=audio 16720 RTP/AVP 0 101
c=IN IP4 172.29.50.186
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

HTH,

Regards,

Mohammed Noor

[Eldho E Kuriakose]

I also heard if the otherside is cordless phone then the problem. Dont know its correct or not

[abdul ilyas]

post traces of successful SIP call ( via mobile).

My guess your gateway is registered to one CUCM & phones are registered to another CUCM server.

[Eldho E Kuriakose]

Phones and gateway registered in same CUCM

My service provider is saying lot of invite messages form my end..

They even asked me to change calling number to 10 digits. That time no call is going through...

[George Thomas]

Can you post debugs after changing the calling number to 10 digits in your DID range? Basically 2903000 should change to a full 10 digit number.

[Eldho E Kuriakose]

That time no national numbers are forwarding..

Please find the logs

calling:0112903000

called:0126496275